1. Field of the Invention
The present invention is particularly applicable to digital processing technology for sounds. The present invention relates generally to adaptive high pass filters, and more particularly to an adaptive high pass filter having a cut-off frequency controllable responsive to a low band signal included in a sound input signal.
2. Description of the Background Art
As examples of a communication system employing a satellite communication, the following systems are known such as a telephone communication system, an answering machine system and a voice mail system. In the satellite communication, a frequency usable for the communication is required to have a narrow bandwidth. A description will be given on sound signal processing applicable to the communication generally known and for other uses.
FIG. 5A is a block diagram of the configuration of a conventional sound signal processing apparatus. Referring to FIG. 5A, this sound signal processing apparatus comprises a microphone 1 which converts a sound to a sound signal, an amplifier 2 which amplifies the sound signal, a high pass filter (hereinafter referred to as HPF) 3c for removing a low band component of the amplified sound signal, a low pass filter (hereinafter referred to as LPF) 4 for removing a high band component of the amplified sound signal, an A/D converter 5, a memory portion 11 for storing the A/D-converted sound signal, and a control portion 6. In addition, this sound signal processing apparatus further comprises, as a circuit for reproducing the sound, a D/A converter 7 which D/A-converts the signal stored in the memory portion 11, an LPF 8 which removes the high band component of the D/A-converted signal, and an amplifier 9 and a loudspeaker 10.
In general, the sound signal is quantized by the A/D converter so as to store the signal into the memory device. When the maximum frequency of the signal component included in an input signal is denoted as f.sub.i(max), a sampling frequency f.sub.s to be used for quantization needs to satisfy the following inequality according to a sampling theorem. EQU f.sub.s .gtoreq.2.multidot.f.sub.i(max) ( 1)
When the sampling frequency f.sub.s is set in the A/D converter if the f.sub.s is not set in accordance with the inequality (1), the signal components of the input signal, which exceed the frequency f.sub.s, is controllable to noise among those involved in the input signal. This noise is known in the art as alias noise. Therefore, as shown in FIG. 5A, the LPF 4 is provided at the preceding stage of the A/D converter to remove the signal components exceeding the frequency f.sub.s of the input signal.
If the sound signal having only the high band components of its input signal cut off by the LPF 4 is A/D-converted, the sound signal obtained by D/A conversion is proceeded to have its low band signal components relatively emphasized. That is, the sound obtained from the sound signal becomes unclear, or becomes poorly articulated. Accordingly, the HPF 3c is provided at the further preceding stage of the A/D converter 5 to obtain a clear or well-articulated sound, so that the low band signal components of the sound signal are also cut off by the HPF 3c.
The operation of the sound signal processing apparatus illustrated in FIG. 5A will now be described briefly. The sound signal obtained from the microphone 1 is amplified by the amplifier 2. The HFP 3c removes the signal components of the frequencies lower than the cut-off frequency from the amplified sound signal. The LPF 4 removes the signal components of the frequencies higher than the cut-off frequency from the output signal of the HPF 3c. The A/D converter 5 A/D-converts the output signal of the LPF 4. The control portion 6 compresses the amount of information corresponding to the signals quantized by the A/D converter 5 to store the compressed data in the memory portion 11.
FIG. 5B shows a useful frequency band limited by the HPF 3c and the LPF 4 in the case of application of the sound signal processing apparatus in FIG. 5A to the telephone system. In this figure, the notation f.sub.CH represents the cut-off frequency of the HPF 3c and is set to approximately 300 Hz. Meanwhile, the f.sub.CL represents the cut-off frequency of the LPF 4 and is set to approximately 3.4 KHz. The cut-off frequencies f.sub.CH and f.sub.CL are both determined by an experiment. In order to prevent an occurrence of alias noise, the signal components exceeding the cut-off frequency f.sub.CL should be removed by band compression. Adaptive Differential Pulse Cord Modulation (hereinafter refereed to as ADPCM) is known as a typical band compression.
When the sound signal including the signal components of 4 KHz or less is processed, for example, the sampling frequency f.sub.S attains 8 KHz. When the sound data is A/D-converted at the speed of 8 bit/sec, for example, a transmission speed of 64 Kbit/sec is required. Thus, due to the application of the ADPCM, the transmission speed of the data to be transmitted is compressed to 32 Kbit/sec.
In reproduction processing of the sound signal, the control portion 6 supplies the data stored in the memory portion 11 to the D/A converter 7 with appropriate decompression if the data has been previously compressed. The data is D/A converted by the D/A converter 7 to be supplied to the LPF 8. The signal fed by the LPF 8 is applied to the amplifier 9 after the high band components are removed thereby. The sound signal amplified by the amplifier 9 is output as a sound via the loudspeaker 10.
FIG. 6 is a block diagram illustrating the configuration of a conventional sound signal processing system. As shown in FIG. 6, two sound signal processing apparatuses are connected via a transmission path 12 such as a digital signal line in the sound signal processing system. Each of these sound signal processing apparatuses is identical to the one whose memory portion 11 is removed from the apparatus shown in FIG. 5A. The FIG. 6 illustrates an example, in which the sound data is not only stored in the memory but also transferred via the transmission path 12.
As shown in FIGS. 5A and 6, the HPF 3c and LPF 4 are provided at the preceding stage of the A/D converter 5 in each of the conventional sound signal processing apparatuses. The LPF 4 has a filter characteristic, which is predetermined in order to prevent the alias noise determined by the sampling frequency f.sub.s of the A/D converter 5. The conventional HPF 3c also has its filter characteristic predetermined. Since the filter characteristic of the HPF 3c is fixed, the low band signal components of the sound signal, depending on only the low frequency band signal components of the input signal, cannot be adaptively removed by the HPF 3c when the signal frequency components included in the sound signal vary depending upon a speaker, a sound field, such as relative positioning of a sound source and a microphone, and microphone characteristic acoustics. For example, the signal frequency components included in the sound signal vary depending on whether the speaker would be male or female. Also, absorption of the low band components of the sound tends to occur in an environment where reverberation easily occurs. In such a case, if removal of the low band components of the sound signal is fixedly performed by the HPF 3 having a fixed filter characteristic, a sound digital signal obtained after A/D conversion is not properly controlled. This causes degradation in a tone quality of the sound obtained by the reproduction processing.
An example of the prior art of particular interest to this invention is seen in Japanese Laying Open No. 62-51827. This art discloses a sound encoding system related to band compression for decreasing the amount of the quantized sound signal to be transmitted. This sound encoding system includes a filter having a variable characteristic, but the characteristic of this filter is not changed responsive to the input signal. In addition, it should be noted that this art is related to the band compression, while the present invention relates to preprocessing for the sound signal.